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SIP/2.0 400 Invalid Contact information RRS feed

  • Question

  • Hello,

     

    I have a problem with incoming SIP calls. Here is my test environement...

     

    MOC2007 <-> OCS2007 <-> Mediation <-> VoIP Gateway

     

    Calls outbound from my MOC2007 client work perfectly, however incoming calls from my VoIP Gateway get returned "SIP/2.0 400 Invalid Contact information". It is actually the OCS2007 server returning this message after receiving the forwarded INVITE from the Mediation server.

     

    Here is the trace.

     

    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:41.866.000001c4 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060<-192.168.195.190:1150
    INVITE sip:+33497234852@192.168.195.114:5060;transport=tcp SIP/2.0
    FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
    TO: <sip:+33497234852@192.168.195.114>
    CSEQ: 4 INVITE
    CALL-ID: 12840797339373401@192.168.195.190
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
    RECORD-ROUTE: <sip:192.168.195.190>
    CONTACT: <sip:0630704615@192.168.195.190;transport=tcp>
    CONTENT-LENGTH: 251
    USER-AGENT: QuesCom SIP Gateway 5.00.007
    CONTENT-TYPE: application/sdp
    ALLOW: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO
    v=0
    o=QuesCom 4308 4308 IN IP4 192.168.195.190
    s=NonSIP
    c=IN IP4 192.168.195.190
    t=0 0
    m=audio 11036 RTP/AVP 8 0 18 101
    a=rtpmap:8 pcma/8000/1
    a=rtpmap:0 pcmu/8000/1
    a=rtpmap:18 g729/8000/1
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    ------------EndOfIncoming SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:41.866.000001d0 (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060->192.168.195.190:1150
    SIP/2.0 100 Trying
    FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
    TO: <sip:+33497234852@192.168.195.114>
    CSEQ: 4 INVITE
    CALL-ID: 12840797339373401@192.168.195.190
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
    CONTENT-LENGTH: 0
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:41.946.0000022e (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883->192.168.195.113:5061
    INVITE sip:+33497234852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;epid=50FADD4DA6;tag=45f6d9ea4
    TO: <sip:+33497234852@testlcs.local;user=phone>
    CSEQ: 11 INVITE
    CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0
    CONTACT: <sipTongue Tiedcottlcs2007med.testlcs.local@testlcs.local;gruu;opaque=srvr:MediationServer:mc3q_s6qrk-5EezN3UoChgAA;grid=d12c4066519c4f8baa1446bac4af551c>;isGateway
    CONTENT-LENGTH: 944
    SUPPORTED: replaces
    SUPPORTED: gruu-10
    USER-AGENT: RTCC/3.0.0.0 MediationServer
    CONTENT-TYPE: application/sdp; charset=utf-8
    ALLOW: UPDATE
    ms-call-source: non-ms-rtc
    ALLOW: Ack, Cancel, Bye,Invite,Refer
    v=0
    o=- 0 0 IN IP4 192.168.195.114
    s=session
    c=IN IP4 192.168.195.114
    b=CT:1000
    t=0 0
    m=audio 62053 RTP/AVP 97 101 115 111 0 8
    c=IN IP4 192.168.195.114
    a=rtcp:63607
    a=candidate:qAIY9y4X7Bw0HEpXWZ98oqsjTz6HlnZC44YyNO/bg+o 1 4QhvRWT/npPPNmNIIUhesw UDP 0.900 192.168.195.114 62053
    a=candidate:qAIY9y4X7Bw0HEpXWZ98oqsjTz6HlnZC44YyNO/bg+o 2 4QhvRWT/npPPNmNIIUhesw UDP 0.900 192.168.195.114 63607
    a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:9kOZ+aKWXlO+Mc5ETULXMUTxW8efQ1j7iiNVlUCC|2^31|1:1
    a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+QL1qybw9F8+wLv7WGLNJAB5R77IyT3rWnZPzpr9|2^31|1:1
    a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:7bEKYRZNsIL4Sd0TqnPz98eHgTLVwETXK2iqnLjs|2^31
    a=label:main-audio
    a=rtpmap:97 RED/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:115 x-msrta/8000
    a=fmtp:115 bitrate=11800
    a=rtpmap:111 SIREN/16000
    a=fmtp:111 bitrate=16000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:43.799.00000253 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883<-192.168.195.113:5061
    SIP/2.0 400 Invalid Contact information
    FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;epid=50FADD4DA6;tag=45f6d9ea4
    TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 11 INVITE
    CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
    VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0;ms-received-port=2883;ms-received-cid=29200
    CONTENT-LENGTH: 0
    ms-diagnostics: 1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"
    ------------EndOfIncoming SipMessage

    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.799.00000262 (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883->192.168.195.113:5061
    ACK sip:+33497234852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;tag=45f6d9ea4;epid=50FADD4DA6
    TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 11 ACK
    CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0
    CONTENT-LENGTH: 0
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.889.000002c5 (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_208E51>], 192.168.195.114:2884->192.168.195.113:5061
    SERVICE sip:+33497234852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:+33497234852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=ca38349310
    TO: <sip:+33497234852@testlcs.local;user=phone>
    CSEQ: 12 SERVICE
    CALL-ID: 36a1168028fd45da882d2b5af58c2ecd
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:2884;branch=z9hG4bK4e403b9a
    CONTENT-LENGTH: 500
    USER-AGENT: RTCC/3.0.0.0 MediationServer
    CONTENT-TYPE: application/msrtc-reporterror+xml
    <?xml version="1.0" encoding="us-ascii"?><reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting"><error callId="bdf869a2-bca4-4ff7-b7d0-022d3709c9ff" toUri="sip:+33497234852@testlcs.local;user=phone" fromTag="45f6d9ea4" toTag="FE355C133B228334B697EF1833317D62" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.889.000002cb (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060->192.168.195.190:1150
    SIP/2.0 400 Invalid Contact information
    FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
    TO: <sip:+33497234852@192.168.195.114>;tag=b82053edfc
    CSEQ: 4 INVITE
    CALL-ID: 12840797339373401@192.168.195.190
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
    CONTENT-LENGTH: 0
    SERVER: RTCC/3.0.0.0 MediationServer
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:44.109.00000322 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_208E51>], 192.168.195.114:2884<-192.168.195.113:5061
    SIP/2.0 401 Unauthorized
    FROM: <sip:+33497234852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=ca38349310
    TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 12 SERVICE
    CALL-ID: 36a1168028fd45da882d2b5af58c2ecd
    VIA: SIP/2.0/TLS 192.168.195.114:2884;branch=z9hG4bK4e403b9a;ms-received-port=2884;ms-received-cid=29300
    WWW-AUTHENTICATE: NTLM realm="SIP Communications Service", targetname="scottlcs2007.testlcs.local", version=3
    WWW-AUTHENTICATE: Kerberos realm="SIP Communications Service", targetname="sip/scottlcs2007.testlcs.local", version=3
    CONTENT-LENGTH: 0
    DATE: Thu, 29 Nov 2007 08:10:38 GMT
    ------------EndOfIncoming SipMessage

     

    Why is the OCS2007 server returning this message ?

     

    How do I correct this ?

     

    Thanks for your help.

     

    Sott

    Thursday, November 29, 2007 9:23 AM

All replies

  • Hi Scott,

     

    Try to have the media gateway present the From number as E.164 with a + prefixed to the Mediation Server. Also make sure you have configured a location profile on the Mediation Server and related normalization rules and have configured your user correctly with msRTCSIP-Line. Have a look in the OCS_VoIP_Guide.

     

    Also I don't think your media gateway is on the list of qualified media gateways http://technet.microsoft.com/en-us/office/bb735838.aspx, so you might run into more problems.

     

    best regards

       Jens

    Thursday, November 29, 2007 10:34 AM
  • I believe I have set up the system correctly, but it still does not work. I now send the + in the FROM field.

     

    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:12.430.00000b65 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060<-192.168.195.190:1087
    INVITE sip:+4852@192.168.195.114:5060;transport=tcp SIP/2.0
    FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
    TO: <sip:+4852@192.168.195.114>
    CSEQ: 1 INVITE
    CALL-ID: 12840810511191070@192.168.195.190
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
    RECORD-ROUTE: <sip:192.168.195.190>
    CONTACT: <sip:+33630704615@192.168.195.190;transport=tcp>
    CONTENT-LENGTH: 253
    USER-AGENT: QuesCom SIP Gateway 5.00.007
    CONTENT-TYPE: application/sdp
    ALLOW: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO
    v=0
    o=QuesCom 18467 18467 IN IP4 192.168.195.190
    s=NonSIP
    c=IN IP4 192.168.195.190
    t=0 0
    m=audio 11010 RTP/AVP 8 0 18 101
    a=rtpmap:8 pcma/8000/1
    a=rtpmap:0 pcmu/8000/1
    a=rtpmap:18 g729/8000/1
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    ------------EndOfIncoming SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.440.00000b71 (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
    SIP/2.0 100 Trying
    FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
    TO: <sip:+4852@192.168.195.114>
    CSEQ: 1 INVITE
    CALL-ID: 12840810511191070@192.168.195.190
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
    CONTENT-LENGTH: 0
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.530.00000bff (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
    INVITE sip:+4852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
    TO: <sip:+4852@testlcs.local;user=phone>
    CSEQ: 21 INVITE
    CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
    CONTACT: <sipTongue Tiedcottlcs2007med.testlcs.local@testlcs.local;gruu;opaque=srvr:MediationServer:mc3q_s6qrk-5EezN3UoChgAA;grid=c60f17471e554939854cdcd79523abfc>;isGateway
    CONTENT-LENGTH: 944
    SUPPORTED: replaces
    SUPPORTED: gruu-10
    USER-AGENT: RTCC/3.0.0.0 MediationServer
    CONTENT-TYPE: application/sdp; charset=utf-8
    ALLOW: UPDATE
    ms-call-source: non-ms-rtc
    ALLOW: Ack, Cancel, Bye,Invite,Refer
    v=0
    o=- 0 0 IN IP4 192.168.195.114
    s=session
    c=IN IP4 192.168.195.114
    b=CT:1000
    t=0 0
    m=audio 61355 RTP/AVP 97 101 115 111 0 8
    c=IN IP4 192.168.195.114
    a=rtcp:62123
    a=candidateSurprisevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 1 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 61355
    a=candidateSurprisevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 2 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 62123
    a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:jJPJeA1RC89pyUK/XPtcyytVP84gRDi5rCgQTq7K|2^31|1:1
    a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:cPvwZ5OF9N9zutdg2JoQIV6XtFpuYULob027AB7w|2^31|1:1
    a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:QjWjKn1QvQIPfhj+ofhp93006LYhIsb6sazBwwjy|2^31
    a=label:main-audio
    a=rtpmap:97 RED/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:115 x-msrta/8000
    a=fmtp:115 bitrate=11800
    a=rtpmap:111 SIREN/16000
    a=fmtp:111 bitrate=16000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:14.393.00000c24 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684<-192.168.195.113:5061
    SIP/2.0 400 Invalid Contact information
    FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
    TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 21 INVITE
    CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
    VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76;ms-received-port=3684;ms-received-cid=29E00
    CONTENT-LENGTH: 0
    ms-diagnostics: 1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"
    ------------EndOfIncoming SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.393.00000c33 (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
    ACK sip:+4852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:+33630704615@testlcs.local;user=phone>;tag=e8952cc14;epid=50FADD4DA6
    TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 21 ACK
    CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
    CONTENT-LENGTH: 0
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.503.00000c5b (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
    SIP/2.0 400 Invalid Contact information
    FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
    TO: <sip:+4852@192.168.195.114>;tag=1ead26a45
    CSEQ: 1 INVITE
    CALL-ID: 12840810511191070@192.168.195.190
    VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
    CONTENT-LENGTH: 0
    SERVER: RTCC/3.0.0.0 MediationServer
    ------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.513.00000cca (S4,SipMessage.DataLoggingHelper:472.idx(500))
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685->192.168.195.113:5061
    SERVICE sip:+4852@testlcs.local;user=phone SIP/2.0
    FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
    TO: <sip:+4852@testlcs.local;user=phone>
    CSEQ: 22 SERVICE
    CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2
    CONTENT-LENGTH: 493
    USER-AGENT: RTCC/3.0.0.0 MediationServer
    CONTENT-TYPE: application/msrtc-reporterror+xml
    <?xml version="1.0" encoding="us-ascii"?><reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting"><error callId="5e80aad3-8fca-4f90-bc1b-db6a9baf35bf" toUri="sip:+4852@testlcs.local;user=phone" fromTag="e8952cc14" toTag="FE355C133B228334B697EF1833317D62" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
    TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:16.796.00000cf3 (S4,SipMessage.DataLoggingHelper:472.idx(550))
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685<-192.168.195.113:5061
    SIP/2.0 401 Unauthorized
    FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
    TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
    CSEQ: 22 SERVICE
    CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
    VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2;ms-received-port=3685;ms-received-cid=29F00
    WWW-AUTHENTICATE: NTLM realm="SIP Communications Service", targetname="scottlcs2007.testlcs.local", version=3
    WWW-AUTHENTICATE: Kerberos realm="SIP Communications Service", targetname="sip/scottlcs2007.testlcs.local", version=3
    CONTENT-LENGTH: 0
    DATE: Thu, 29 Nov 2007 11:50:14 GMT
    ------------EndOfIncoming SipMessage

    Thanks again for your help.

    Scott

     

    Thursday, November 29, 2007 1:25 PM
  • Same problem on my OCS <-> VOIP Gateway settlement. Calling from Communicator to Gateway users run fine; every call originated by gateway users receive SIP/2.0 400 Invalid Contact Information. Which is the Normalization rule role in inbound OCS calls ? I have a location profile normalization rule: ^\+(\d{5})$  <--> $1  for manage outbond calls, it's checked for inbound calls too ?
    This is the SIP log: (Every attempt with (+) in INVITE/FROM/TO header was done. Same result)


    INVITE sip:44653@mediation.tsf.local;transport=tcp SIP/2.0

    Record-Route: <sip:10.57.3.141:5080;lr;sipX-route=%2Afrom%7EYzE0N2RiMWI%60.authrules%2Aauth%7E%21822274c8a0ce72388ed7b8363600c832>

    Max-Forwards: 18

    Contact: <sip:+200@10.57.7.20:17216>

    To: "+44653"<sip:+44653@10.57.3.141>

    From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b

    Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.

    Cseq: 1 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

    Content-Type: application/sdp

    User-Agent: X-Lite release 1011s stamp 41150

    Content-Length: 412

    Date: Wed, 09 Jan 2008 10:26:18 GMT

    Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b

    Via: SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c

    Via: SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216



    v=0

    o=- 2 2 IN IP4 10.57.7.20

    s=CounterPath X-Lite 3.0

    c=IN IP4 10.57.7.20

    t=0 0

    m=audio 5006 RTP/AVP 107 119 100 106 0 105 98 8 101

    a=alt:1 1 : UTFJi2Ky +xViCroo 10.57.7.20 5006

    a=fmtp:101 0-15

    a=rtpmap:107 BV32/16000

    a=rtpmap:119 BV32-FEC/16000

    a=rtpmap:100 SPEEX/16000

    a=rtpmap:106 SPEEX-FEC/16000

    a=rtpmap:105 SPEEX-FEC/8000

    a=rtpmap:98 iLBC/8000

    a=rtpmap:101 telephone-event/8000

    a=sendrecv

    SIP/2.0 100 Trying

    FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b

    TO: "+44653"<sip:+44653@10.57.3.141>

    CSEQ: 1 INVITE

    CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.

    VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216

    CONTENT-LENGTH: 0



    SIP/2.0 400 Invalid Contact information

    FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b

    TO: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be

    CSEQ: 1 INVITE

    CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.

    VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer



    ACK sip:44653@mediation.tsf.local;transport=tcp SIP/2.0

    Contact: sip:+200@10.57.7.20:17216

    From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b

    To: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be

    Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.

    Cseq: 1 ACK

    Max-Forwards: 20

    Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b

    Content-Length: 0




    Wednesday, January 9, 2008 11:10 AM
  • Hi Alberto, has this issue been resolved? I'm experiencing exactly the same issue (posted yesterday) where I can't call from my Mitel IP phone to the Office Communicator client. Had no responses yet, so was wondering if your problem is already solved.

     

    Thanks

     

     

    Friday, January 18, 2008 11:20 AM
  • Issue is still open for me. Maybe SIP/TCP stack of our VOIP Gateway is buggy/lack but I'm not be able to make calls from Dialogic Gateway to OCS mediation server work too.
    Monday, January 21, 2008 3:28 PM